asterisk users - Feb 2008 - Error checking asterisk method - suggestions?

Hi there dear users and dear developers of Asterisk!


I've got a maybe strange idea, let's see if you laugh or think it's
reasonable J

I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
Each analog line can be reached through a phonenumber, since they are each
connected to my telephone provider. Yes expensive I know J


The challenge:

I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines,
and 3) my operator is alive and working, and I have an Idea which I wonder if
will/could work.

My first idea was to ask the zap-driver if it could detect if the line was ok,
but no function existed to do that, what I could find. Anyone knows about one?

My second idea, was to try calling simply, to know if things were ok. But, I
couldn't just call any number, I had to know the number was in use, and not
disturbing anyone.
So, I called myself, or I called another of my phonelines.

So,
I'd like to use the asterisk manager interface in java to originate a call
from one ZAP-channel, calling out to my telephone provider,
And then they will direct the call back to my, but into another ZAP-channel
(since I'm calling that channel's number).

So: I'm making ZAP/1 calling out to no 323121321 -> telephone company,
Ok: 323121321 belongs to this guy -> redirecting me to my ZAP/2 channel,
which answers the call.

Then I have a connection, and ZAP/2 will answer and do some DTMF.
My first ZAP/1 is run through my java program, and I'd like to listen for
certain DTMF-tones, to know I have a working circuit.

The goal for all of this, is to verify things are working, so my provider is not
down, or one of my ZAP-lines are dead.

So far, I've tried calling and got some half-success, but not sure what is
going on doing all the right way.
For ex: why am I calling with Zap1, to Zap 3, and then Zap 7 is answering? 3
channels used for one outgoing and one incoming call? Something must be very
wrong J
Please educate me, dear experts.

Input?


Sincerely,
Johan Sandgren
www.svep.se<http://www.svep.se>, jsa at svep.se
Frosty Sweden but with some sunshine today !! J

My code and settings below, for information.

=============JAVA CODE (extract)

OriginateAction originateAction = new OriginateAction();
ManagerResponse originateResponse = null;
originateAction.setChannel("ZAP/1");
originateAction.setContext("Outgoing");
originateAction.setExten("201");                                 //
maps to ZAP/7 through external phonecompany PBX
originateAction.setPriority(new Integer(1));
originateAction.setTimeout(new Long(15*1000));                                  
// xml-milliseconds
originateAction.setAsync(false);

============== extensions.conf (extract)

[Incoming]
exten => s,1,Set(DYNAMIC_FEATURES=hangup)
exten => s,2,Agi(agi://localhost/answer.agi)

[Outgoing]
exten => _X.,1,Set(DYNAMIC_FEATURES=hangupfeature)
exten => _X.,n,Dial(Zap/3/${EXTEN}

==============  Asterisk response:

AGI Debugging Enabled
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'stt' logged on from 127.0.0.1
       > Channel Zap/1-1 was answered.
    -- Executing [201 at Outgoing:1] Set("Zap/1-1",
"DYNAMIC_FEATURES=hangupfeature") in new stack
    -- Executing [201 at Outgoing:2] Dial("Zap/1-1",
"Zap/3/201") in new stack
    -- Called 3/201
    -- Starting simple switch on 'Zap/7-1'
    -- Zap/3-1 answered Zap/1-1
[Feb 14 13:02:33] WARNING[26260]: chan_zap.c:6499 ss_thread: CallerID returned
with error on channel 'Zap/7-1'
    -- Executing [s at Incoming:1] Set("Zap/7-1",
"DYNAMIC_FEATURES=hangup") in new stack
    -- Executing [s at Incoming:2] AGI("Zap/7-1",
"agi://localhost/answer.agi") in new stack
AGI Tx >> agi_network: yes
AGI Tx >> agi_network_script: answer.agi
AGI Tx >> agi_request: agi://localhost/answer.agi
AGI Tx >> agi_channel: Zap/7-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1202990552.2
AGI Tx >> agi_callerid: unknown
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: unknown
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: Incoming
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 2
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << ANSWER
AGI Tx >> 200 result=0
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=6
AGI Rx << WAIT FOR DIGIT 10000
  == Manager 'testmanager' logged off from 127.0.0.1
AGI Tx >> 200 result=0
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=6
AGI Rx << WAIT FOR DIGIT 10000
AGI Tx >> 200 result=0
  == Spawn extension (Incoming, s, 2) exited non-zero on 'Zap/7-1'
    -- Hungup 'Zap/7-1'

========= Asterisk log

__________________________________________________________________
Johan Sandgren
Svep Design Center AB (www.svep.se<http://www.svep.se>)
St. Lars v?g 42A, SE-222 70 Lund
Phone: 046-19 27 22

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On Thu, Feb 14, 2008 at 01:17:45PM +0100, Johan Sandgren
wrote:
> Hi there dear users and dear developers of Asterisk!
> 
> 
> I've got a maybe strange idea, let's see if you laugh or think
it's reasonable J
> 
> I'm using Asterisk with Digium TDM800P cards with 24 lines (as an
answering machine).
> Each analog line can be reached through a phonenumber, since they are each
connected to my telephone provider. Yes expensive I know J
> 
> 
> The challenge:
> 
> I'd like to somehow verify that my 1) TDM800P cards and 2) the analog
lines, and 3) my operator is alive and working, and I have an Idea which I
wonder if will/could work.
> 
> My first idea was to ask the zap-driver if it could detect if the line 
> was ok, but no function existed to do that, what I could find. Anyone 
> knows about one?
What do you mean by "line is OK"?
> 
> My second idea, was to try calling simply, to know if things were ok. 
> But, I couldn't just call any number, I had to know the number was in 
> use, and not disturbing anyone.
> So, I called myself, or I called another of my phonelines.
And you assume noone else calls in at the time?
> 
> So,
> I'd like to use the asterisk manager interface in java to originate a 
> call from one ZAP-channel, calling out to my telephone provider,
> And then they will direct the call back to my, but into another 
> ZAP-channel (since I'm calling that channel's number).
For a basic test that the line works, try TestClient and TestServer .
Originate a call from testclient (and set there the number. And set all
incoming calls temporarily to go to TestServer (did I mention the
assumption that noone calls in?)

One relatively cheap method of "temporary" is through setting a global
variable to a non-standard value. This means that the non-default value
will never last after a reload. And you can set the global even through
'core set global VARNAME VALUE' in the CLI.

Check the resulting reports in /var/lib/asterisk/testresults . Make sure
all of them were generated, and that none "FAIL"-ed.

-- 
               Tzafrir Cohen
icq#16849755              jabber:tzafrir.cohen at xorcom.com
+972-50-7952406           mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir